Which parts of SIP packet are consuming a packet size? SDP?

Using $b2bua=true reduces the size of the SIP packet.

It seems you already put it in the DeployPatterns.
but do you still have the issue?

Try DialPlan rule like this.

[Matching Patterns]
$request = ^REGISTER
$addrtype = 6
$param(Via,"received") = \[(.+)\]
$removeparam(Via,"received") = (.+)

[Deploy Patterns]
Via = %2;received=%1
$continue = true

"net.bind.use" enables the interface binding.

> Would I need to set net.bind.interface explicitly before using net.bind.use=true?

It doesn't matter.

Add the line below in the [Configuration]->[Advanced] page. It might meet the requirement.

It is logged in Error Log because $response is usually used for sending an error response code.

Because the SIP Server sends "200 OK" to OPTIONS, it meets the requirement.

So you don't have to worry about it.

How do I import User Authentication and what does the imported correct file in csv format look like when opened via excel? Yes. A CSV file can be edited with Excel. Is it possible to change the ip address and network segment of the client device on the Brekeke SIP server? Is it an IP address shown ...

The DialPlan rules you wrote return "400 Bad Request" to the 3CX and the PSTN.. Try the rule below. This rule accepts an INVITE without the authentication if it is sent from 3CX or PSTN Matching Patterns $request = ^INVITE $addr = ^3CX_IP$|^PSTN_IP$ Deploy Patterns $auth = false $continue = true ...

I don't recommend you to disable the authentication for INVITE, what you did with DialPlan, if the SIP Server can be reached from the Internet. I recommend you to check the remote IP address of 3CX/PSTN with $addr in DialPlan to avoid someone using your SIP Server freely. https://docs.brekeke.com ...

Yes, you can replace these SIP headers with DialPlan.

For example..
[Matching Patterns]
$request = ^INVITE
$getSIPuser(Remote-Party-ID) = 1(.+)

[Deploy Patterns]
From = sip:%1@192.168.1.181:5060

What SIP UA client product are you using?
Does the second INVITE sent to UA3 have a different Call-ID from the original INVITE sent to UA2?

Have you restarted the SIP Server after you changed the configuration?
Do you still have the issue?

Let you set [Peer Certification Validation] = "off" in [Configuration]->[SIP] page.

Do you need to replace Oracle Java 8 with AdoptOpenJDK 11? The wiki topic below will help you. https://docs.brekeke.com/sip/upgrade-java-version-to-java-11-or-later But, to avoid downtime, I recommend that you prepare another Windows machine to install AdoptOpenJDK 11 and Brekeke SIP Server latest ...

It seems there is no RFC which defines " vad=no".

Hi h.fabien and o.mahmoud, Did you add new DialPlan rules or modify default rules? You need to keep the default "From PBX" and "To PBX" rules applied to bridge SIP calls for WebRTC client such as JsSIP. If you added any new DialPlan rules, let you disable them to apply default rules.

Try GMT+4 and restart the SIP Server.

As a SIP proxy, BSS passes SDP what Avaya sent. so BSS doesn't care even if there are multiple codecs.

I recommend you to talk to Avaya admin to configure the trunk because the settings are depending on versions and deployment.

FYI:
https://docs.brekeke.com/interop/config ... pbx-cm-5-2

If a firewall is running, disable it or open the UDP/TCP port 5060.
Also the SIP Server process should have a privilege to open the port.

> 1. Brekeke Product Name and Version: 3.3.5.8 Advanced Update it to the latest version because 3.3.5.8 is pretty old. https://docs.brekeke.com/sip/sip-history > 1. When we click on the Registered Clients tab I get this error: Are you using a Thirdparty database? Did you set any value at ...

set &net.nat.force=false in Deploy Patterns.

Do you have the one-way audio issue with other calls too?

Let you look at the SIP Server's log...